similar to: Re: Asterisk-Users Digest, Vol 8, Issue 216

Yes - the problem was a missing signalling line in zapata.conf. Now in and out work. Also, it was news that "reload" from the console doesn't reflect changes made in zaptel and zapata.conf entries. Any other config files that it doesn't reload? >From: "Asterisk@Home" <asteriskathome@yahoo.com> > >Did you go into AMP and configure some place for

If I try to reshape a data frame into a long format with more than one column it seems to mix up the column headings (or the columns, depending on how you look at it). For example: > d <- data.frame(row=1:2,b.1=c("1b1","2b1"),b.2=c("1b2","2b2"),a.1=c("1a1","2a1"),a.2=c("1a2","2a2")) > d row b.1 b.2 a.1 a.2 1

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

Sender: EMBROIDERED <tsl2u.c0shr@myweb.cn> Reply-To: meilung1931@yahoo.com.tw Date: Thu, 16 Dec 2004 12:08:13 +0800 X-Mailer:Dynamailer V 8.4 X-MimeOLE:Produced By Mircosoft MimeOLE V6.00.2600.0000 Return-Path:meilung1931@yahoo.com.tw [r-devel@stat.math.ethz.ch ] EMBROIDERED PATCHES CUSTOM POLICE , FIRE, SPORTS , CLUBS

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Jeez, I need to work out the shortcut to send an email which I keep pressing by accident!! -----Original Message----- From: C. Tomlinson [mailto:asterisk_list@burntwires.com] Sent: 27 February 2005 22:48 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DISA and a long delay; ideas? Many thanks, that was the problem. I didn't paste the

I was getting MANY emails from XXX.hinet.net - kind of filled my mqueue directory. I stopped sendmail. removed all the files in mqueue, added to sendmail access a REJECT for hinet.net This certainly seems to help - but is that the correct way to take care of this sort of thing? Thanks, Jerry

Hello list: I have dom0 that is supported PPP in kernel and i have installed ppp tools for dailing-up. I have a question , How should i use ppp dail-up in domU ??? First, I should recompile domU kernel thus support ppp ,right??? How''s next ???? Please help.... -- axa@HiNet <axanet@ms32.hinet.net> _______________________________________________ Xen-users mailing list

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

[moved from -dev, as the thread is better suited for -users] At 5:10 PM -0600 1/30/04, James Sharp wrote: > > I've pretty much got the routing covered at this point, I'm just not sure >> how to get the Asterisk system to answer and give me dialtone immediately. >> Any ideas or recommendations would be greatly appreciated. > >app_disa will give answer and give

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my

I can dial from an asterisk host to another one via FreeWorldDialup, on the other side DISA service answer to me and i can ear dialtone. But i cannot send DTMF and dial an extension on the DISA enabled asterisk.....i've tried rfc2833 and inband...but nothing....any tips ??? Thanks, -- Igor Barsanti GPG Public key available at http://pgp.mit.edu

When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against the context that the channel is associated with from the configuration in zapata.conf. With

I am a little green when it comes to all this but I am trying to connect our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able to dial an extension on my PBX handset and I get a dialtone from the PBX. After 2 rings I then hear the asterisk server connect and I get a dialtone from asterisk. I am then able to dial an extension on another asterisk server. My question is: How do

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>